The Advanced Mobile Phone System (AMPS) is a mature partially analog frequency-modulated (FM) mobile communications system, which is in wide use throughout the United States. Widely used AMPS handsets employ analog FM quadrature discriminator and modulator technology. These analog AMPS handsets must demodulate three primary sub-signals. These sub-signals include an analog voice signal, a supervisory audio tone (SAT) and wideband data. The downlink or received analog voice signal is compressed using simple mu-law compression yielding 1 dB gain changes for every 2 dB input change. The analog voice signal passes through a pre-emphasis filter typically having a 6 dB/octave response. The analog voice signal is restricted to a phase deviation of 12 kHz and bandlimited between 300 Hz and 3000 Hz. Thus, the analog voice signal has a peak phase deviation .beta..sub.voice of no more than 2.9 radians. The SAT provides a means of identifying a base station occupying a cellular channel. The SAT is always present in the AMPS signal and must be continually transponded. The SAT, therefore, must be isolated with the downlink and added to the uplink or transmitted signal. It is a fixed-frequency sinusoid of 5970 Hz, 6000 Hz or 6030 Hz. The SAT has a peak frequency deviation of 2 kHz yielding a peak phase deviation .beta..sub.SAT of no more than 1/3 radians. Wideband data is the only digital part of the AMPS signal. It is 10 kbps (kbits-per-second) Manchester encoded data. The two possible signal levels (+1, -1) yield a frequency deviation of 8 kHz. This wideband data is never simultaneously present with the analog voice signal. A dotting sequence of alternating ones and zeros precedes the wideband data control messages. This dotting sequence presents a strong spike at 5 kHz after FM demodulation.
An AMPS voice signal for carrier signal.function.c may be represented as EQU A(t)=cos(2.pi..function.ct)-sin(2.pi..function.ct).phi.(t) [1]
The voice signal of equation [1] may be simplified where .phi.(t) is close to zero so that at intermediate frequency .function..sub.IF the voice signal is represented as EQU AIF(t)=cos(2.pi..function.IFt+.phi.(t)) [2]
where .phi.(t) is a complex argument. This intermediate frequency .function..sub.IF passes through a limiter, thereby producing a signal based on its sign. This limited IF signal (almost a square wave) is represented as EQU z(t)=sgn{AIF(t)} [3]
and is presented to a typical analog discriminator.
These analog discriminators, however, require substantial area and have filter bandwidths and phase distortion that are difficult to control. Moreover, if digital processing is required, an antialiasing filter and analog-to-digital converter (ADC) must follow the analog discriminator. Similarly, AMPS transmit requirements such as pre-emphasis, companding and other signal processing functions such as integration and filtering are costly to implement with analog designs.